I saw this updated “news” about migrating to pjsip since chan_sip is deprecated https://www.freepbx.org/freepbx-migration-towards-pjsip/
but how do we handle the NAT issue? I cant tell you how many phones I have using chan_sip because the NAT feature is not working in pjsip. Maybe I am missing something but in chan_sip I have to set the NAT to YES (forced) for my remote phones to work. Every time I tried pjsip I can not get these phones to work and I dont see the NAT option in the pjsip settings page??? What am I missing?
Someone else will likely chime in on where and what they are in FreePBX, but from a general PJSIP perspective it has the same functionality for NAT handling as chan_sip except it uses separate options instead of a singular “nat” option. Rewrite Contact, Symmetric RTP, Force RPort, Direct Media. There are also options on the transport in case Asterisk itself is behind NAT.
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ok, I will do some searching in the GUI.
Thanks.
Couldn’t find NAT in extension settings for PJSIP, any suggestions for Grnadstream gxp2170. I dont want to set this in EPM as a basefile edit. I am looking for a setting in the extension. I can not get pjsip to work remotely through some ISP’s, chan_sip works fine.
As @jcolp said, RTP Symmetric, Rewrite Contact, Force rport and Direct Media all default to on and (except for the last which is very unlikely be be active) correspond to forcing NAT on chan_sip.
If you are using default settings (pjsip listening on 5060 and chan_sip on 5160), then a SIP ALG at either the remote site or the PBX could be causing your issue.
If you want technical help, paste a log (including SIP trace) of a failed registration or call at pastebin.freepbx.org and post the link here.
I confirmed all those settings are on. If SIGALG is the issue then cant I just change my PJSIP port to a nonstandard one on both ends?
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